[Freeswitch-users] 30 Second call drop.

[Paul]

Thanks for the suggestion Anthony, I will check the logs again to see if
there is any TIMEOUT errors... (I don't think there was).


On Thu, July 18, 2013 7:15 am, Anthony Minessale wrote:
> Hi,
>
>
> Is it only during voicemal?  If so, Try this:
>
>
> http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
>
>
> Otherwise turn on sip trace with "sofia global siptrace on" and "console
> loglevel debug" and post it.
>
>
>
>
>
> On Thu, Jul 18, 2013 at 10:54 AM, Paul <pasha at prosperity4ever.com> wrote:
>
>
>> Hi guys,
>>
>>
>> After hours of googling and trying every different config options I
>> need some help!
>>
>> My scenario is as follows:
>>
>>
>> Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun)
>> Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
>>
>>
>> Main Network (where Freeswitch resides): 10.0.0.0/24
>>
>>
>> My sip gateway/proxy: 10.0.0.40 (kamailio)
>> Freeswitch: 10.0.0.34
>>
>>
>> I can fully use the VPN from both networks (ssh, web gui control, etc)
>>
>>
>> phones can register, however here are my problems:
>>
>> When calling incoming DID none of the extensions ever get ring, it
>> skips to voicemail... when leaving a message voicemail cuts of after
>> 30-31
>> seconds exactly. I am wondering if freeswitch isn't detecting my client
>> sending audio and interprets it as silence and therefore hangs up. The
>> voicemail message itsel does contain everything said right up to the
>> hang up.
>>
>> Sometimes I can call between extensions if they are on the same side of
>>  the VPN subnet (say 101 calling 101 and they both are coming from
>> 192.168.1.0/24 via 10.8.0.0/24)
>> If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says
>> user is unavailable.
>>
>> I'm suspecting this is a NAT issue, all the reading I have done talks
>> about "external sip and rtp ip" in my case my trunk hooks up from
>> private ip to private ip, it's the kamailio that's supposed to do the
>> NAT (which I
>> have verified works fine, I can place outgoing calls through it from
>> other PBXs no problem with two way audio and working as it should be).
>>
>>
>> I'm not sure if there is a way to enable nat or not with freeswitch.
>> One
>> thing I have done (as I've found this worked for some people) is set all
>>  references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34
>> (my
>> freeswitch eth0). There are no other adapters on the box.
>>
>> I am hoping that as this is my first time setting up freeswitch that
>> one of you freeswitch experts can point me in the right direction :)
>>
>> Thanks in advance guys!
>>
>>
>>
>>
>> _______________________________________________________________________
>> __
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org http://www.freeswitchsolutions.com
>>
>>
>> 
>> 
>>
>>
>> Official FreeSWITCH Sites
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>>
>>
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>>
>>
>
>
>
> --
> Anthony Minessale II
>
>
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>
>
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>
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> _________________________________________________________________________
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>
>
> 
> 
>
>
> Official FreeSWITCH Sites
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> http://www.cluecon.com
>
>
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